LCOV - code coverage report
Current view: top level - libavformat - audiointerleave.c (source / functions) Hit Total Coverage
Test: coverage.info Lines: 60 73 82.2 %
Date: 2017-12-17 23:02:56 Functions: 4 4 100.0 %

          Line data    Source code
       1             : /*
       2             :  * Audio Interleaving functions
       3             :  *
       4             :  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
       5             :  *
       6             :  * This file is part of FFmpeg.
       7             :  *
       8             :  * FFmpeg is free software; you can redistribute it and/or
       9             :  * modify it under the terms of the GNU Lesser General Public
      10             :  * License as published by the Free Software Foundation; either
      11             :  * version 2.1 of the License, or (at your option) any later version.
      12             :  *
      13             :  * FFmpeg is distributed in the hope that it will be useful,
      14             :  * but WITHOUT ANY WARRANTY; without even the implied warranty of
      15             :  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
      16             :  * Lesser General Public License for more details.
      17             :  *
      18             :  * You should have received a copy of the GNU Lesser General Public
      19             :  * License along with FFmpeg; if not, write to the Free Software
      20             :  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
      21             :  */
      22             : 
      23             : #include "libavutil/fifo.h"
      24             : #include "libavutil/mathematics.h"
      25             : #include "avformat.h"
      26             : #include "audiointerleave.h"
      27             : #include "internal.h"
      28             : 
      29          14 : void ff_audio_interleave_close(AVFormatContext *s)
      30             : {
      31             :     int i;
      32          39 :     for (i = 0; i < s->nb_streams; i++) {
      33          25 :         AVStream *st = s->streams[i];
      34          25 :         AudioInterleaveContext *aic = st->priv_data;
      35             : 
      36          25 :         if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
      37          12 :             av_fifo_freep(&aic->fifo);
      38             :     }
      39          14 : }
      40             : 
      41          14 : int ff_audio_interleave_init(AVFormatContext *s,
      42             :                              const int *samples_per_frame,
      43             :                              AVRational time_base)
      44             : {
      45             :     int i;
      46             : 
      47          14 :     if (!samples_per_frame)
      48           0 :         return AVERROR(EINVAL);
      49             : 
      50          14 :     if (!time_base.num) {
      51           0 :         av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
      52           0 :         return AVERROR(EINVAL);
      53             :     }
      54          39 :     for (i = 0; i < s->nb_streams; i++) {
      55          25 :         AVStream *st = s->streams[i];
      56          25 :         AudioInterleaveContext *aic = st->priv_data;
      57             : 
      58          25 :         if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
      59          36 :             aic->sample_size = (st->codecpar->channels *
      60          24 :                                 av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
      61          12 :             if (!aic->sample_size) {
      62           0 :                 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
      63           0 :                 return AVERROR(EINVAL);
      64             :             }
      65          12 :             aic->samples_per_frame = samples_per_frame;
      66          12 :             aic->samples = aic->samples_per_frame;
      67          12 :             aic->time_base = time_base;
      68             : 
      69          12 :             aic->fifo_size = 100* *aic->samples;
      70          12 :             if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
      71           0 :                 return AVERROR(ENOMEM);
      72             :         }
      73             :     }
      74             : 
      75          14 :     return 0;
      76             : }
      77             : 
      78        1456 : static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
      79             :                                        int stream_index, int flush)
      80             : {
      81        1456 :     AVStream *st = s->streams[stream_index];
      82        1456 :     AudioInterleaveContext *aic = st->priv_data;
      83             :     int ret;
      84        1456 :     int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
      85        1456 :     if (!size || (!flush && size == av_fifo_size(aic->fifo)))
      86        1235 :         return 0;
      87             : 
      88         221 :     ret = av_new_packet(pkt, size);
      89         221 :     if (ret < 0)
      90           0 :         return ret;
      91         221 :     av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
      92             : 
      93         221 :     pkt->dts = pkt->pts = aic->dts;
      94         221 :     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
      95         221 :     pkt->stream_index = stream_index;
      96         221 :     aic->dts += pkt->duration;
      97             : 
      98         221 :     aic->samples++;
      99         221 :     if (!*aic->samples)
     100         172 :         aic->samples = aic->samples_per_frame;
     101             : 
     102         221 :     return size;
     103             : }
     104             : 
     105        1289 : int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
     106             :                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
     107             :                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
     108             : {
     109             :     int i, ret;
     110             : 
     111        1289 :     if (pkt) {
     112         784 :         AVStream *st = s->streams[pkt->stream_index];
     113         784 :         AudioInterleaveContext *aic = st->priv_data;
     114         784 :         if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
     115         501 :             unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
     116         501 :             if (new_size > aic->fifo_size) {
     117           0 :                 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
     118           0 :                     return AVERROR(ENOMEM);
     119           0 :                 aic->fifo_size = new_size;
     120             :             }
     121         501 :             av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
     122             :         } else {
     123             :             // rewrite pts and dts to be decoded time line position
     124         283 :             pkt->pts = pkt->dts = aic->dts;
     125         283 :             aic->dts += pkt->duration;
     126         283 :             if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
     127           0 :                 return ret;
     128             :         }
     129         784 :         pkt = NULL;
     130             :     }
     131             : 
     132        3740 :     for (i = 0; i < s->nb_streams; i++) {
     133        2451 :         AVStream *st = s->streams[i];
     134        2451 :         if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
     135        1235 :             AVPacket new_pkt = { 0 };
     136        2691 :             while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
     137         221 :                 if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
     138           0 :                     return ret;
     139             :             }
     140        1235 :             if (ret < 0)
     141           0 :                 return ret;
     142             :         }
     143             :     }
     144             : 
     145        1289 :     return get_packet(s, out, NULL, flush);
     146             : }

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